The Chicken And The Egg

In which I crack open the shell of a standard filter and show you the insides.
Reaktor 6 (affiliate link)
tinyurl.com/mr2uesd8
EQ Doesn't Cause Phase Shift...
• EQ Doesn't Cause Phase...
Robert Bristow Johnson Audio EQ Cookbook:
webaudio.github.io/Audio-EQ-C...
Music: "Egghead" by Dan Worrall.
Bandcamp link: dan-worrall.bandcamp.com/trac...
If you like this type of content and you want to see it more often, consider signing up for Channel Membership: / @danworrall
Or you can just buy me a coffee: ko-fi.com/danworrall

[@donit.]

Can't wait for the full C++ course next video because one commenter thinks the phaseshift is caused by the Reaktor implementation.

[@glebglub]

which ends on a cliffhanger because we have to THEN explore how c++ is being compiled on the assembly level so we can make our OWN version of C with minor tweaks and improvements, called DanC

[@disklamer]

@@glebglub Follow-up will be building auro-optical hardware to prove minor fluctuations in lightspeed do not affect your hearing

[@shanonkiyoshi4784]

​@@glebglub DanC? Or Dancy?

[@shanonkiyoshi4784]

​@@disklamer No creature on the planet can HEAR light 😂🤣

[@disklamer]

@@shanonkiyoshi4784 See, that”s where we disagree 😅

[@sebastiaanbarends]

Even though he forces me to use my brain again, I'm glad my favorite teacher is back from sick leave.

[@Schlingelkind]

2:07 thanks Dan for going back to that time to provide actual proof. You put a lot of effort in for us!! ❤

[@wagayuuri9397]

Another great video explaining DSP basics. I find it incredible how people will just start arguments over things they clearly haven't been taught and surely don't understand, but insist they know better than what signal processing engineers have been doing for literally more than half a century. In fact, if anyone wants to really get into the itty bitty nits and details of DSP from the ground up, I can only recommend the MIT OpenCourseWare playlist of lectures on Digital Signal Processing, held by Alan Oppenheim... from 1975. And this is still exactly the same material being taught taught today. Because it works.

[@edgriggs3544]

Math - it works, bitches.

[@riibs6166]

It makes me sad when Dan tells his viewers to send hate comments in the chat. Your music is awesome! Keep up the great work! ❤

[@alexhulme6135]

That egg photo cracked me right up! Thanks for the god tier videos as always

[@Beatsbasteln]

dude, at this point your youtube channel has become a full on dsp education course. holy shit! great work

[@martinweber988]

I'm a simple man. I get a bandcamp notification about a new music release by Dan Worrall and I go looking for a new video. My nerd senses tingle in anticipation.

[@bluberrykush3912]

I will quote that last sentence regularly from now on😂

[@kpeleent]

I just love your sense of humor.

[@marcorademan8433]

Hi Dan, I would like to provide my input on "phase shift causes EQ". I am currently a PhD student student specializing in signal processing, so I speak from a mathematical perspective. In electronic circuits, phase and amplitude have a direct relationship. EQ is a result of the circuit design, which may utilize the anplitude-phase relationship to varying degrees. Sometimes phase is more important in some designs, while others focus more on the relationship between both. In DSP, phase shifts may be viewed as "causing" amplitude shifts in the case IIR filters (filter output is a function of filter input). In FIR fIlters, filter output is simply a "rolling weighted sum" of the filter input. It is the FIR impulse response that effects the signal, and phase is simply a by-product of the shape of the impulse response. Finite IRs may be optimized to provide certain characteristics, such as minimum phase, or maximum filter steepness for a given FIR time support at the expense of phase, or linear phase at the expense of FIR length and filter steepness. In the case of EQ, one may utilize a linear combination of a phase shifted signal to approximate nearly any amplitude response, but it not the only way of achieving this goal. Unfortunately, it is a complex topic, and the relationship between phase and amplitude (or "EQ") is inevitably intertwined, which is highly dependent on the application, or whether it is a physical circuit or digital.

[@ghfjfghjasdfasdf]

Howdy, I recently started using EQ in my productions. FabFilter 3’s eq on top of a couple other filters in a chain caused my rendered tune to “come off”. I’ve always rendered a bit of extra space at the end of a tune in my daw, to ensure that I don’t cut off any of the tail, and it’s always been easy to trim the silence at the end via my wave editor. After utilizing FabFilter 3 (simply a smidge of low shelf around 150 hz @ +5 DB) on top of a couple other filters (an instance of Live’s Autofilter running a HP and an instance of one of Palmary’s filters, a dual LP setup doing all kinds of automation), I noticed that the end of my tune was off of center all the way to the end. I noticed zero audible artifacts, tune sounded ok… it just looked ugly compared to what I’m used to. Does this sound par for the course when it comes to using EQ? Thanks

[@marcorademan8433]

@@ghfjfghjasdfasdf Hi! It is the DAWs job to pass sample buffers (list of audio samples) to the plugins. The buffers typically have a power of 2 length (256, 512 etc). The plugins will then process the audio provided in the buffers and pass them back to the DAW for further processing. Although I am uncertain of how this works under the hood, I would imagine that plugins may require various buffer lengths, as some processing (e.g., very long filters or IRs) cannot be computed without longer buffers. I would assume that the DAW will compensate for each specific plugin delay, in order to properly sync the audio during playback. As such, a long chain of effects will inevitable cause a sample delay, which is likely the shift you observe in your final render. On a side note, if the DAW did not compensate for plugin buffer delays, mixing tracks will result in a mess of notch filters created everywhere (if you add a signal to a time-shifted variant of itself, it forms a notch filter). So I would think that the DAW will compensate for individual plugin delays, thereby shifting the entire master track as a result to match the longest delay in a processing chain. In ProQ3, I am confused by what they mean with their "zero latency" mode, as audio buffers are, by definition, not zero latency. However, I assume "zero-latency" simply means that a single buffer goes in, and immediately comes out processed, without the need for waiting for additional buffers and other processing-delay causes. When using IIR filters, it is also possible to immediately provide an output given a short length of input samples, given very low latency indeed. This is not the case for FIR filters (which includes all IRs such as cabinet emulators and convolutional reverbs). FIR filters require a buffer length of at least the length of the IR in order to compute a single output sample! I hope this gives you some insight into your observation! Also, please take my answer with a grain of salt, as it is only an educated guess. I am not a DAW programmer ;)

[@ghfjfghjasdfasdf]

I truly appreciate you taking the time to respond in depth, it is much appreciated. For yiur information, I had the zero latency option selected. Have a care! 🤟

[@beefeeb]

@@ghfjfghjasdfasdfit kinda sounds like you’re seeing a DC offset?

[@ghfjfghjasdfasdf]

Thank you SO much!! ^ That’s exactly what it is… shows my ignorance. It’s the first time I’ve ever produced a wave that looked like that. Now I need to figure out how I caused it. Can’t thank you enough for taking the time to chime in. 🫡

[@natigrinkrug]

Now THIS is true content. This is exactly the stuff I want to see on my KZbin recommended page, and I can't say I expected anything less from the excellent content Dan Worrall has been putting out for the past few years. Great stuff, man, I think I speak for the majority of us when I say that we really love what you do :)

[@zavireshiran9841]

thank you for the link to the audio eq cookbook. i've been looking for something like that for years now

[@neofaizan]

Imagine doubting this man.

[@szundymusic]

I love that “photographic proof” !!! 😂❤

[@yelpo6388]

I remember my first digital audio class and finding out that all the effects are delays, in our field we have to add 1 meter to each step so that they appear 15cm in perspective.

[@michaelm1]

I bow to the master. I feel like a caveman listening to a modern, high-level quantum physicist. So much to learn, so little time.

[@eyeball226]

I love it when you get into super nerdy stuff. I do a bit of (very) amateur JSFX plugin development, so thank you!

[@sonidojamon]

2:00 As proven in Caveman ("starring" Ringo Starr), fried eggs happened before chickens. That movie also is a testament to how music was created. Anyone thinking humans didn't co-exist with dinosaurs is wrong as well. If this movie isn't proof enough, check the Cadillacs & Dinosaurs comics and animated series!

[@bassplayingchris]

For some reason I loved those pictures so much lol. Can't wait to check Tokyo dawns new offerings after work!!

[@markhadman]

One of your best. I've been very slowly (re-)learning calculus since 2020, with linear algebra up "next" - and one day I hope to have a good handle on DSP, so this video hits home for me.

[@glebglub]

power to ya buddy! if you want some university-grade physics knowledge on sound/vibrations check out Brock University's PPLATO section Phys 5 - Vibrations and waves! there's also a lot of maths stuff at the bottom, starting from basic algebra all the way up to solving 2nd order differential equations (such as in harmonic damping) and waves/partial differentiation (i.e. Schrodinger equation). probably overboard but you may find something useful. I'd give a direct link but I don't want to risk it knowing how youtube is with links lol

[@DerpySwag]

i fear for the day dan worrall gets 20 million subs, and he gets dragged into audio engineering DRAMA, and has to own all the haters with just straight facts and logic so sound in knowledge and reality that they literally would have zero things to say against it i fear if this man has any enemies, this guy is like the john wick of audio. i can't describe it..

[@shanonkiyoshi4784]

He totally IS the JOHN WICK of Audio!!! 🎶💥😎👍💥🎶

[@NPC_averagemale003]

I just have to say that your content is the most outstanding, scientific and educational videos I've ever seen on KZbin in the field of audio engineering. All your objective tests fulfills the scientific standards in methods of measurement whether it be visual, aural or numerical. Thank you for being an absolute truth in this hypercapitalistic industry ridden with snake oil business practices, paid promoters, visual eye candy, religion, cognitive biasness and genuinely superstitous individuals that prey on the newcomers as well as cannibalize amongst themselves.

[@frejberglind7271]

To me, first order low pass filters are the most intuitive, since they are basically just a moving average. You can create an all pass filter using a first order low pass like this: f(x) = x - 2 * lpf(x). I find this a very natural way of understanding an all pass filter.

[@umbrachris2455]

this tune is awesome Dan

[@orbitfold]

Introduction To Digital Filters with Audio Applications by Julius O Smith gives a great and thorough introduction to filters even for mathematically „unsophisticated“

[@dompetro]

Dan Worrall is the reason I come to KZbin everyday. A lovely gent he is. And smart too!

[@marshy_moo]

Dan Worrall = Honey Badger (gets Reaktor NFR but then uses freeware equivalent! 😂).

[@macronencer]

Really interesting insights, thank you. I was especially interested to learn why dynamic and modulated filters use more CPU. Fascinating stuff!

[@Kabelsprak]

Brilliant 😊 Love your stuff, keep it up man!

[@morgan0]

yea the phase shift comes first, from the unit delay. but that fixed phase shift has to be combined with feedback to be more directly useful. z transforms are both complicated and simple at the same time, converting H(z) to Y(n) is some simple replacement and rearranging, but constructing H(z) with proper coefs is more complicated, and how that transform works is more complicated. i learned it a while back to make my own custom peak/bell filters and while it was cool to get a deeper understanding of filters, it was also nice to be done and move on to other things lol. maybe i’ll revisit it in the future to make more custom filters, who knows, i certainly always have ideas.

[@gughffhhghgghghgg1690]

Help me out a bit here,i dont understand much of this cause im no eq expert,but in the end,can you explain me in a simple way if eq causes phase shift ot the other way around,and also if the phase shift is a result of sample delay?

[@morgan0]

@@gughffhhghgghghgg1690 so there’s a few methods that come to mind that could be used for eq, most used is minimum phase, where i would say the phase shift causes the frequency change. in my original comment i said that a unit delay is a frequency dependent phase shift, and that’s the building block of minimum phase or analog style digital filters (in actual analog filters, the phase response comes from the frequency dependent phase shift in the current across a capacitor. next is linear phase, which only works because its non-causal, you need to know data from the future to decide the current sample, and as a result the whole sound is delayed, but evenly. FIR filters like this can be represented as a large number of non-feedback configured unit delays, so it’s possible the phase shift is still the cause, albeit indirectly. and thirdly (but likely not lastly), is fourier transform processing, this requires a buffer of samples to process, but i don’t think _requires_ delay, but i think typically is delay compensated. in this case samples from the block are mathed together (i’d have to refresh my memory on how it works exactly) to get a limited number of frequency bins, which can have their amplitudes and phases modified independently. however if we take a unit delay to be phase shift, then indirectly still phase shift comes first. the unit delay creates a phase shift of 0 degrees at 0 hz and 180 degrees at +-nyquist.

[@clampchowder9569]

wow I watched the first one, then saw this one in the recommended, just a video called the chicken and the egg, and i wanted to see what it was about so i queued it, I was so bamboozled that it was a sequel to the first one, then I listened to the background music which I really enjoyed and then checked out this guys spotify and its some really great stuff. Wild journey.

[@makeaguitarnoise]

I've got reactor 6.great it is. Your music is great Dan. Original with no commercial potential 😂

[@ItsWesSmithYo]

Best low key swing at the knees in the business 😂

[@Mind_Idiot]

Excellent work as always Dan.

[@ghfjfghjasdfasdf]

Dan, How do you find the time to make awesome music, make extremely educational youtube videos, and win the volume wars taboot? Legend

[@originaali]

Old school type dubstep 🔥 ace

[@Tony-Stockport]

My brain hurts. But I'm happy. What did I learn from Dan in this video? That this stuff is way above my comprehension level. Thank you Mr Worrall. Even though I'll fail this class you're still a cool teacher.

[@ElrondMcBong.]

3:17 the fact that this is a terminal text file made me so happy I just love efficiency so much! :D

[@lyghters]

Terminal text file? You mean "plain" text?

[@sonidojamon]

Can't get enough of internet trolls getting obliterated by Mr Worrall. Dan, I hope you're doing better health-wise, by the way!

[@andrigunawan5691]

But can we make a parametric equalizer using same integration methods as in typical game physics engine, but at audio-rate since mass-spring system is basically an IIR resonant lowpass filter?

[@theappearedone]

Now thats what ive been waiting for, thanks

[@bitegoatie]

I like "Egghead" - so sorry for the positive reaction - despite the naughties wobble (you deploy the wobble in a subtle, heady way). I like the title, as well. Well played, educator.

[@saftpackerl]

Great stuff! Please program an Alesis Midiverb/II with Reaktor now!

[@rhinoskin7550]

You've exposed me to something I never even thought about haha I'm such a noob. This is so much knowledge

[@jankalinowski6767]

Nice Sounds in the Intro!!

[@SianaGearz]

You can calculate the coefficients once per frame (say 512 samples or what have you) and interpolate them across the frame, to solve the problem that changing coefficients becomes computationally heavy. Or i'm sure a lot of implementations don't even interpolate, though that can be a little perilous. Assuming the input parameters are twiddled at a human speed and not at a sample speed.

[@Am6-9]

Sample speed would be unlikely, as (apart from vst3) no plugin format allows for sample accurate automation/parameter changes, afaik those changes are usually reported once per every process call (so every n samples, depending on the buffer size) VST3 theoretically allows for sample accurate automation, but a) my guess is that very few plugins implement that and b) it would be insane to do so…

[@SianaGearz]

@@Am6-9 back when MIDI was a wire standard, you couldn't have much more than a thousand messages per second.

[@amptechron]

Amazing as usual!

[@karvn1148]

Hey Dan love the vids mate. I was wondering if u had any pointers for bass levels and bass translation across speakers. E.g. bass level on headphones sounds fine, but way too boomy and overpowering in the car.

[@DanWorrall]

It's all about accurate monitoring. Your headphones aren't telling you the truth about your bass frequencies. I would suggest better headphones, or maybe some EQ correction. I don't usually mix on headphones, but I stopped being surprised / shocked by the bottom end of my mixes when I first got some decent full range monitors and did a bit of room treatment.

[@brokko_le3]

Any chance on a Cenozoix compressor review? Would love your take on it 🥰

[@bontempo1271]

Hope you're keeping well Dan !

[@DerpySwag]

great video as always

[@05degrees]

Yeah, math is such that seemingly different _semantics_ (and not just computation algorithms) can be in the end sides of the same coin. This can be found in all branches and applications, DSP not excepted. Making an all-pass with adding two parallel band-pass filters and making a band-pass by mixing pure signal and all-passed signal are also an instance of having a space which admits different bases, and there is no one preferred basis so we can think a thing can be decomposed into A and B, but likewise it can be decomposed into C and D, even if that thing is A itself (or C, etc.). Cases of multifaceted math in physics do even relate to some misconceptions people have about physics. For example, general relativity can be formulated both as a theory of pure geometry of curved spacetime _and_ as a theory of a field on a _flat_ spacetime, which isn’t well-known by many but appears in in-depth textbooks. Then people argue about gravity “not being a force” and denying themselves an additional way to think about it, just because they thought there’s only one way. This can even be linked back to sound: decomposition of a signal into sinusoidal partials (or better, imaginary exponents) isn’t the “most natural” one and there are no Platonic sines propagating everywhere-it’s just very useful. But sometimes it isn’t (for example, wavelets are of some use). In fact, using Fourier analysis by itself can’t reproduce some of our sound perception quirks: AM of a white noise (of any distribution) with a pulse-width wave makes it sound more or less tonal when Fourier image of this signal doesn’t get any peaks at the corresponding frequency or its harmonics.

[@soundcore183]

In summary, the allpass filter is a useful tool for manipulating the phase characteristics of a signal while maintaining a constant magnitude response. It finds applications in various audio processing tasks, such as creating artificial reverberation or phase correction in equalization.

[@duroxkilo]

useful in crossover setups too

[@soundcore183]

@@duroxkilo multiband configurations, reverbs, phasers

[@duroxkilo]

@@soundcore183 weddings, bridal showers, etc :):) jokes aside, i'm a bit fascinated w/ the topic of sound processing. i wish i had the brains to understand the math behind digital filters but i can barely keep up w/ the basics of 'analog sound'..

[@BrunodeSouzaLino]

I wonder how much of the "commentsphere" will break when the "Reverb is EQ" video comes out.

[@ulfrohdin]

At first I did not agee. But then there was photographic proof 😂

[@AndreaCipria]

I see a Dan Worrall video, I INSTANTLY have to open it.

[@ChuckNoise]

I understod every Word of this amazing video - when i Saw it for the 20th time :D

[@drrodopszin]

Your videos give me some crazy ideas always, but now I'm not sure if I just reinvented EQ or not 😄. I made a parallel channel to specifically target a hissing distorted guitar frequency with MFreeShift. So I selectively changed the phase in one area. Then it's just EQ, no?

[@racebase1470]

great video dan as always , thanks a lot ! i Know its out of context here but i have a question i wanted to ask you while ago , when i finish a project after mix and master i make sure the dynamics is closed to my reference similar track and the overall loudness is closed , its sounds good on my monitors and headphones and most systems. In the phone at middle volume its sounds good, but in full volume its like i puted 20 limiters and the mix is flat and lifeless. and the reference track sounds clean ive tested if im doing something wrong with my master and after checking - instagram audio or spotify its sounds quite clean loud and punchy at full volume but when i open the track trough soundcloud or the files folder, its sounds crushed in terms of dynamics at higher volumes. - the question - is modern phones speaker cannot pull out audio accurate at full volume when the track is louder than -14 lufs ? and what In your opinion could cause the problem if not this 🙏

[@francobuzzetti9424]

hey Dan! can you explain clippers? and how they are or aren't just fast AF compressors?

[@user-xj6xf5bf3j]

WHATEVER YOU SAY GOES DAN!

[@Imperceptible_parachute]

As BoC said: Music is Math. I will sleep well tonight. Thanks, Dan!

[@gh0stransist0r]

nightmare for an art student. Going to stick with Mountain Spirit Guru and everything it blessed me with.

[@siabeats7422]

Hi dan! Love your work. I think we all can agree minimun phase EQs have a post ring and linear phase one have the pre ring as well because the impulse response creates it and the higher the slope the bigger the impulse response and the more ringing we have. But! In your way of designing it seems there are no need for impulse responses. Is that mean if we design EQs like you there will be no more pre or post ringing??

[@DanWorrall]

No. Allpass filters also ring.

[@siabeats7422]

​@DanWorrall oh! Thanks a lot for clearing that one for me.

[@myceliumbug]

me at the middle of that reaktor chain explaination: "uhhhhhhh ...you are completly right, my man!"

[@ir8123]

I'm expecting that next video would be about how different DAW's sounds differently 🔥🔥

[@glebglub]

that's easy to find out, just import the .exe as raw data into audacity and hit play! ;D

[@glebglub]

hey Dan, do you have any recommendations for a free tracker instrument/soundfont player? I have tons of .XI files from the KIArchive years ago, but the plugin I used to use (Jeskola XS-1) is 32-bit only, and the only good paid ones I can find cost so much I might as well just buy FL Studio. and all the free ones (that I can find at least), are either terribly buggy or lack basic features like an ASDR override also, nice track. would have fit nicely in a dubstep rave back int' day

[@weedwick9888]

I like your music

[@bearwithit]

Next video: The Birds and The Bees

[@bradleyduer]

I recently bought Volcano 3, pretty much, solely because of your videos. I wanted to use your affiliate link, but couldn't find one. I'm thinking about buying the Total Bundle, but I'd really like to do it with your link. I wouldn't be purchasing their products if not for you. If you happen to see this, can I get your FF affiliate link?

[@scottarmstrong5346]

EQ GO BRRRRRRR

[@tjpaeldon1]

The Gordon Ramsey of Audio🤷🏽‍♂️😆🍻

[@No.0.o.0]

I knew what you meant about getting an upgrade to 6 but i mean they have been neglecting their synths in favor of hundreds of kontakt instruments and everything else.

[@jotunbjorn]

Is this just a thing in the digital world? Do analog filters just send unwanted frequencies to ground or do they affect phase as well?

[@DanWorrall]

Analogue filters also affect phase, and acoustic filters. Real life is minimum phase.

[@jotunbjorn]

@@DanWorrall Thanks for the reply. Guess I'll have to do a deep dive.

[@rafriley6993]

One thing I don’t understand is; when you apply the eq boost and subsequent opposite cut, showing in plugin doctor that the phase has effectively returned back to normal, does that mean that every sound has some kind of inherent correct phase? Like if I were to drag in a sample that I’d EQ’d previously, would that phase shift be detectable by some kind of analyser? And therefore could I ‘straighten’ out a samples phase issues with eq? If that makes sense

[@DanWorrall]

There's no inherent correct phase, no. If you move a microphone closer to or further from the source, that will shift the phase of higher frequencies more than lower frequencies.

[@RegebroRepairs]

Yeah, but you can use that same math to make an all pass filter from a standard filter. Basically, filters and phase change are two parts of the same thing. It's a but like saying subtracting is just adding.

[@paulricketts10]

Unless you do a null test I'm not interested!

[@darbomusic]

Love it 😂😂😂

[@JohnOverturf]

Big brained DSP gurus are my best friends

[@barnem108]

how do i upload songs on instagram without the subs sounding distorted and wobbly? the one question i cant seem to find answer to im assuming theres multiple factors

[@DanWorrall]

No idea, don't do Instagram. Does the platform use loudness normalising, and if so what kind? Does it stream at a low bitrate, or downmix to mono or something nasty like that?

[@DmitryPuffin]

Instagram reduces the quality of the songs if they are added to reels/stories. You can try to render your song in lower bitrate and check how it sounds. As a solution, you might need to set ceiling of your limiter a bit lower, like -0.3db. That helps to retain quality during file converting, as your signal doesn't hit 0 and there is a headroom. But that's just what I do, and that's might be far from being correct 😅 i just learned that from other producers.

[@DanWorrall]

@@DmitryPuffin I would need more specific details to offer any suggestions. Does reduced quality just mean very low bitrate? Or do they also reduce the samplerate, or drop to mono etc?

[@DmitryPuffin]

@@DanWorrall Instagram uses 128kbps 44.1khz AAC stereo, as far I know. Of course it might sound better than mp3 with same specs, but still its lossy. Also, instagram has app settings where it might turn video quality down if network connection is weak. As per my experiments, that also might affect the quality of content that is processed by app on the phone and uploaded that way, meaning that if automatic quality adjustment is enabled, that might lead to worse quality material upload when app decides that network is weak. So low end and high end will suffer in this case comparing to the wav file exported from the DAW.

[@barnem108]

to be honest i aint giot a clue what its doing i all i know my tunes sound worst in the low end than before but i hear other peoples tunes they sound bang on. just thought is ask on this page thank you @@DanWorrall

[@LorneVignettes]

Hi Dan.

[@yaayimanalien]

wowzer

[@GillamtheGreatest]

I understood almost none of that but still feel smarter.

[@MOSMASTERING]

Serious question... how much would it cost to just sit next to you for a week? We could even mix a track! 😂 It was always my dream since I was a teen in the 90s to work as a teaboy in a studio! Just to be near a mix engineer and learn from a master is a trend that is going the way of the Dodo - those opportunities just don't exist any more. Everywhere you look there are home studios and all the big studios are gone . One of my favourite artists of all time is Simon Posford (Shpongle, Younger Brother) his tracks in the genre with which he works just blows everything else out the water - turns out he learned the trade by working in studios around engineers all the time.

[@DaveChips]

Me running like a kid towards ice cream van... Papa Dan, papa Dan... Can we get a video on Cenozix compressorr? 😊😁 They are claiming that it's a first compressor with ADDA anti aliasing algorithm.

[@KlaraKopf]

The single cell is first. After this the whole organism is build. _Final answer!_ 🙂

[@sssyntax]

Howdy Dan, hey I'm a big fan of your music and I was wondering if you would make a video about understanding and comprehending the illusion of audio? How to create certain audio illusions and why they work the way they do. In painting and drawing, a rocky mountain is brown and grey up close but becomes purple the further away it gets. Nothing is what it appears to be. Like in the Simpson's "Uh, Sir, why don’t you just use real cows? Cows don’t look like cows on film. You gotta use horses. What do you do if you want something that looks like a horse? Ehh, usually we just tape a bunch of cats together" Audio seems to be just the same and I havnt heard anyone talk about or explain how the illusion of audio works.

[@DanWorrall]

It's all illusions really! Arguably I cover that in most videos. Eg, the illusion that you're hearing multiple separate sources when in fact it's one stereo waveform. We create that illusion with the arrangement, the balance and the EQ. Or the illusions of width and depth when we're only actually hearing two point sources: any time I talk about stereo techniques.

[@JohnPaulJapes]

🤯 I love your stuff but this is proper brain exploding! 🤣

[@patricknaughton9322]

The egg was around for millions of years before the chicken..

[@benstanfill363]

This was cool, but I was hoping I'd hear your take on what came first, the chicken or the egg.

[@goodtimejohnny8972]

🤯

[@Studio22mix]

This un understandablablable 😵‍💫

[@modellking]

To speed up trig functions, programmers commonly use LUTs

[@PocinTheTech]

Hi Dan, I wanted to let you know that I made a video about your chicken and egg problem that may explain, why there is a bit of confusion about phase shift and EQs. I would feel honoured, if you could check it out and tell me, if I may have gotten something wrong there.

[@Schlingelkind]

Arghh, I hate that music, angrrryyy face!!! Hope you're happy now, grrr! Jikes aside, I love your videos

[@bjornkameleon]

Phase Math 202

[@jupiterman7]

I'm just a simpleton, I just wanna make interesting sounds. If it sounds good it is good 😂

[@1loveMusic2003]

Before I watched this video I considered myself a fairly smart fella but now I have doubts.

[@FeJotaTakinOva]

Just because of that dinosaur pic you got my like 😂 you got me laughing for a while with that one.

[@sleepCircle]

it's ai art, i'm 99% sure

[@DanWorrall]

You can bump that up to 100%

[@sleepCircle]

@@DanWorrall remember, you're still paying for "credits" and things on these website, and this was all trained on stuff taken without consent from artists who are already typically insanely underpaid. midjourney developers, for example, were caught bragging about all the specific, named artists' styles they've stolen and carefully 'laundered' to avoid detection. they've managed to make the acquisition of "custom art" cheaper, but only by obscuring the connection between the product they're selling you, and the hundreds of millions of man hours-a majority percentage of which are being struggled out by people still alive-which made its existence possible. sort of like how people eat the chocolate bar and don't see the child slavery because the connection is so distant and vague; obfuscated by machinery and numbers. a monetary vote for ai art is a monetary vote for, somewhere, some regular artist's extinction.

[@DanWorrall]

@@sleepCircle this is clearly nonsense. If ai art were not available I wouldn't have commissioned a human artist to paint that picture for me. And the joke would not have been so funny if I had.

[@sleepCircle]

@@DanWorrall i think maybe youtube only shows you the original comment in your alerts, not any subsequent edits. i addressed the price thing

[@M1ster77]

I must remember not to eat wasabi drenched sushi while watching these videos ... it does not go well having laughing fits caused by Dans' humour and eating wasabi...😱🤣